Gabe, > If my setup goes: Phone => asterisk => asterisk => PSTN termination provider > Can I define "canreinvite" on both asterisk boxes so the phone call will go > directly to the PSTN provider?
Yes, you can reinvite multiple times. The media path will collapse as much as possible. It works reliably, unless the two servers are too close to each other. Don't ask my why. My server is 1.2 ms RTT away from my provider's server and on about 50% of the calls I end up with a "482 Loop Detected" response to my reinvite on incoming calls. I found that putting a Ringing entry in my dialplan and a 0.5 second delay before Answer fixes it. I tried tracking this down but didn't have much luck. It's fine on outgoing calls. --Luki _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
